It is well documented that the delay, introduced by digital processing in modern audio systems, can lead to a range of disturbing effects experienced by the user. The processing delay should in general be lower than 10 milliseconds. This time is based on average ratings and rather large deviations exist depending on degree of: amplification, incoming sound signal, type of sound processing and individual differences between people. The range of acceptable values may be roughly 3 to 40 milliseconds depending on such factors.
While a short delay is desirable in order to limit the disturbing effects experienced by the user, (poor sound quality, difficulty in locating direction of sound source) when a short delay is specified it severely limits the processing capabilities of a given audio system.
Hence, the more advanced processing used in the system, the longer the delay will inevitably be. One example is noise reduction oriented processing which is often based on block processing, and if the system is only allowed to impose a short delay, only very limited block length can be used leading to poorer performance.
In state of the art audio systems a certain fixed processing delay is imposed. This delay is a compromise between the risk of subjectively experienced problems and the processing capabilities.
In connection with audio devices of the hearing aid type there has been a trend in recent years towards more open hearing aids, i.e. instruments with large vent diameters. Such open instruments may be particularly sensitive to the delay introduced by the audio processing. At the same time there is a push for more time consuming signal processing features enhancing the wanted signal (typically a speech signal).
According to the disclosure of US 20020122562 A1, there exists many possible tradeoffs between the number of bands, the quality of the bands, filterbank delay and power consumption. In general, increasing the number or quality of the filterbank bands leads to increased delay and power usage. For a fixed delay, the number of bands and quality of bands are inversely related to each other. On one hand, 128 channels would be desirable for flexible frequency adaptation for products that can tolerate a higher delay. The larger number of bands is necessary for the best results with noise reduction and feedback reduction algorithms. On the other hand, 16 high-quality channels would be more suitable for extreme frequency response manipulation. Although the number of bands is reduced, the interaction between bands can be much lower than in the 128 channel design. This feature is necessary in products designed to fit precipitous hearing losses or other types of hearing losses where the filterbank gains vary over a wide dynamic range with respect to each other. In accordance with the invention presented in the US 20020122562 document, the filterbanks provide a number of bands, which is a programmable parameter.
The US document does not allow the change of processing time to be performed on-line during processing, but solely mentions the possibility to program a certain delay or frequency resolution prior to the use of the audio device. Thus the user will have to live with this programmed setting, even if the audio environment changes and changes in processing in terms of more time delay and more complex processing would suddenly be advantageous.
The invention provides a method of audio processing and an audio device which offers a solution to this problem.